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https://github.com/FFmpeg/FFmpeg.git
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8e576d5830
* qatar/master: libavutil: add utility functions to simplify allocation of audio buffers. libavutil: add planar sample formats and av_sample_fmt_is_planar() avconv: fix segfault at EOF with delayed pictures pcmdec: remove unneeded resetting of samples pointer avconv: remove a now unused parameter from output_packet(). avconv: formatting fixes in output_packet() avconv: declare some variables in blocks where they are used avconv: use the same behavior when decoding audio/video/subs bethsoftvideo: return proper consumed size for palette packets. cdg: skip packets that don't contain a cdg command. crcenc: add flags avconv: use vsync 0 for AVFMT_NOTIMESTAMPS formats. tiffenc: add a private option for selecting compression algorithm md5enc: add flags ARM: remove needless .text/.align directives Conflicts: doc/APIchanges libavcodec/tiffenc.c libavutil/avutil.h libavutil/samplefmt.c libavutil/samplefmt.h tests/ref/fate/bethsoft-vid tests/ref/fate/cdgraphics tests/ref/fate/film-cvid-pcm-stereo-8bit tests/ref/fate/mpeg2-field-enc tests/ref/fate/nuv tests/ref/fate/tiertex-seq tests/ref/fate/tscc-32bit tests/ref/fate/vmnc-32bit Merged-by: Michael Niedermayer <michaelni@gmx.at>
373 lines
12 KiB
C
373 lines
12 KiB
C
/*
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* Copyright (c) 2010 S.N. Hemanth Meenakshisundaram
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* Copyright (c) 2011 Mina Nagy Zaki
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* memory buffer source for audio
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*/
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#include "libavutil/audioconvert.h"
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#include "libavutil/avstring.h"
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#include "libavutil/fifo.h"
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#include "asrc_abuffer.h"
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#include "internal.h"
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typedef struct {
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// Audio format of incoming buffers
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int sample_rate;
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unsigned int sample_format;
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int64_t channel_layout;
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int packing_format;
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// FIFO buffer of audio buffer ref pointers
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AVFifoBuffer *fifo;
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// Normalization filters
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AVFilterContext *aconvert;
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AVFilterContext *aresample;
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} ABufferSourceContext;
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#define FIFO_SIZE 8
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static void buf_free(AVFilterBuffer *ptr)
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{
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av_free(ptr);
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return;
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}
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static void set_link_source(AVFilterContext *src, AVFilterLink *link)
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{
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link->src = src;
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link->srcpad = &(src->output_pads[0]);
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src->outputs[0] = link;
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}
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static int reconfigure_filter(ABufferSourceContext *abuffer, AVFilterContext *filt_ctx)
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{
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int ret;
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AVFilterLink * const inlink = filt_ctx->inputs[0];
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AVFilterLink * const outlink = filt_ctx->outputs[0];
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inlink->format = abuffer->sample_format;
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inlink->channel_layout = abuffer->channel_layout;
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inlink->planar = abuffer->packing_format;
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inlink->sample_rate = abuffer->sample_rate;
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filt_ctx->filter->uninit(filt_ctx);
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memset(filt_ctx->priv, 0, filt_ctx->filter->priv_size);
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if ((ret = filt_ctx->filter->init(filt_ctx, NULL , NULL)) < 0)
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return ret;
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if ((ret = inlink->srcpad->config_props(inlink)) < 0)
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return ret;
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return outlink->srcpad->config_props(outlink);
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}
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static int insert_filter(ABufferSourceContext *abuffer,
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AVFilterLink *link, AVFilterContext **filt_ctx,
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const char *filt_name)
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{
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int ret;
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if ((ret = avfilter_open(filt_ctx, avfilter_get_by_name(filt_name), NULL)) < 0)
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return ret;
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link->src->outputs[0] = NULL;
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if ((ret = avfilter_link(link->src, 0, *filt_ctx, 0)) < 0) {
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link->src->outputs[0] = link;
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return ret;
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}
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set_link_source(*filt_ctx, link);
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if ((ret = reconfigure_filter(abuffer, *filt_ctx)) < 0) {
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avfilter_free(*filt_ctx);
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return ret;
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}
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return 0;
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}
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static void remove_filter(AVFilterContext **filt_ctx)
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{
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AVFilterLink *outlink = (*filt_ctx)->outputs[0];
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AVFilterContext *src = (*filt_ctx)->inputs[0]->src;
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(*filt_ctx)->outputs[0] = NULL;
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avfilter_free(*filt_ctx);
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*filt_ctx = NULL;
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set_link_source(src, outlink);
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}
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static inline void log_input_change(void *ctx, AVFilterLink *link, AVFilterBufferRef *ref)
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{
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char old_layout_str[16], new_layout_str[16];
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av_get_channel_layout_string(old_layout_str, sizeof(old_layout_str),
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-1, link->channel_layout);
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av_get_channel_layout_string(new_layout_str, sizeof(new_layout_str),
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-1, ref->audio->channel_layout);
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av_log(ctx, AV_LOG_INFO,
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"Audio input format changed: "
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"%s:%s:%d -> %s:%s:%d, normalizing\n",
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av_get_sample_fmt_name(link->format),
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old_layout_str, (int)link->sample_rate,
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av_get_sample_fmt_name(ref->format),
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new_layout_str, ref->audio->sample_rate);
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}
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int av_asrc_buffer_add_audio_buffer_ref(AVFilterContext *ctx,
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AVFilterBufferRef *samplesref,
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int av_unused flags)
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{
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ABufferSourceContext *abuffer = ctx->priv;
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AVFilterLink *link;
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int ret, logged = 0;
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if (av_fifo_space(abuffer->fifo) < sizeof(samplesref)) {
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av_log(ctx, AV_LOG_ERROR,
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"Buffering limit reached. Please consume some available frames "
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"before adding new ones.\n");
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return AVERROR(EINVAL);
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}
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// Normalize input
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link = ctx->outputs[0];
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if (samplesref->audio->sample_rate != link->sample_rate) {
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log_input_change(ctx, link, samplesref);
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logged = 1;
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abuffer->sample_rate = samplesref->audio->sample_rate;
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if (!abuffer->aresample) {
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ret = insert_filter(abuffer, link, &abuffer->aresample, "aresample");
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if (ret < 0) return ret;
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} else {
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link = abuffer->aresample->outputs[0];
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if (samplesref->audio->sample_rate == link->sample_rate)
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remove_filter(&abuffer->aresample);
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else
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if ((ret = reconfigure_filter(abuffer, abuffer->aresample)) < 0)
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return ret;
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}
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}
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link = ctx->outputs[0];
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if (samplesref->format != link->format ||
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samplesref->audio->channel_layout != link->channel_layout ||
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samplesref->audio->planar != link->planar) {
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if (!logged) log_input_change(ctx, link, samplesref);
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abuffer->sample_format = samplesref->format;
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abuffer->channel_layout = samplesref->audio->channel_layout;
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abuffer->packing_format = samplesref->audio->planar;
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if (!abuffer->aconvert) {
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ret = insert_filter(abuffer, link, &abuffer->aconvert, "aconvert");
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if (ret < 0) return ret;
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} else {
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link = abuffer->aconvert->outputs[0];
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if (samplesref->format == link->format &&
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samplesref->audio->channel_layout == link->channel_layout &&
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samplesref->audio->planar == link->planar
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)
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remove_filter(&abuffer->aconvert);
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else
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if ((ret = reconfigure_filter(abuffer, abuffer->aconvert)) < 0)
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return ret;
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}
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}
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if (sizeof(samplesref) != av_fifo_generic_write(abuffer->fifo, &samplesref,
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sizeof(samplesref), NULL)) {
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av_log(ctx, AV_LOG_ERROR, "Error while writing to FIFO\n");
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return AVERROR(EINVAL);
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}
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return 0;
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}
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int av_asrc_buffer_add_samples(AVFilterContext *ctx,
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uint8_t *data[8], int linesize[8],
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int nb_samples, int sample_rate,
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int sample_fmt, int64_t channel_layout, int planar,
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int64_t pts, int av_unused flags)
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{
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AVFilterBufferRef *samplesref;
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samplesref = avfilter_get_audio_buffer_ref_from_arrays(
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data, linesize, AV_PERM_WRITE,
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nb_samples,
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sample_fmt, channel_layout, planar);
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if (!samplesref)
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return AVERROR(ENOMEM);
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samplesref->buf->free = buf_free;
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samplesref->pts = pts;
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samplesref->audio->sample_rate = sample_rate;
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return av_asrc_buffer_add_audio_buffer_ref(ctx, samplesref, 0);
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}
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int av_asrc_buffer_add_buffer(AVFilterContext *ctx,
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uint8_t *buf, int buf_size, int sample_rate,
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int sample_fmt, int64_t channel_layout, int planar,
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int64_t pts, int av_unused flags)
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{
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uint8_t *data[8];
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int linesize[8];
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int nb_channels = av_get_channel_layout_nb_channels(channel_layout),
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nb_samples = buf_size / nb_channels / av_get_bytes_per_sample(sample_fmt);
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av_samples_fill_arrays(data, linesize,
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buf, nb_channels, nb_samples,
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sample_fmt, 16);
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return av_asrc_buffer_add_samples(ctx,
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data, linesize, nb_samples,
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sample_rate,
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sample_fmt, channel_layout, planar,
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pts, flags);
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}
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static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
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{
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ABufferSourceContext *abuffer = ctx->priv;
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char *arg = NULL, *ptr, chlayout_str[16];
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char *args = av_strdup(args0);
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int ret;
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arg = av_strtok(args, ":", &ptr);
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#define ADD_FORMAT(fmt_name) \
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if (!arg) \
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goto arg_fail; \
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if ((ret = ff_parse_##fmt_name(&abuffer->fmt_name, arg, ctx)) < 0) { \
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av_freep(&args); \
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return ret; \
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} \
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if (*args) \
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arg = av_strtok(NULL, ":", &ptr)
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ADD_FORMAT(sample_rate);
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ADD_FORMAT(sample_format);
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ADD_FORMAT(channel_layout);
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ADD_FORMAT(packing_format);
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abuffer->fifo = av_fifo_alloc(FIFO_SIZE*sizeof(AVFilterBufferRef*));
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if (!abuffer->fifo) {
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av_log(ctx, AV_LOG_ERROR, "Failed to allocate fifo, filter init failed.\n");
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return AVERROR(ENOMEM);
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}
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av_get_channel_layout_string(chlayout_str, sizeof(chlayout_str),
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-1, abuffer->channel_layout);
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av_log(ctx, AV_LOG_INFO, "format:%s layout:%s rate:%d\n",
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av_get_sample_fmt_name(abuffer->sample_format), chlayout_str,
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abuffer->sample_rate);
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av_freep(&args);
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return 0;
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arg_fail:
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av_log(ctx, AV_LOG_ERROR, "Invalid arguments, must be of the form "
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"sample_rate:sample_fmt:channel_layout:packing\n");
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av_freep(&args);
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return AVERROR(EINVAL);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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ABufferSourceContext *abuffer = ctx->priv;
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av_fifo_free(abuffer->fifo);
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}
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static int query_formats(AVFilterContext *ctx)
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{
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ABufferSourceContext *abuffer = ctx->priv;
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AVFilterFormats *formats;
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formats = NULL;
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avfilter_add_format(&formats, abuffer->sample_format);
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avfilter_set_common_sample_formats(ctx, formats);
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formats = NULL;
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avfilter_add_format(&formats, abuffer->channel_layout);
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avfilter_set_common_channel_layouts(ctx, formats);
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formats = NULL;
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avfilter_add_format(&formats, abuffer->packing_format);
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avfilter_set_common_packing_formats(ctx, formats);
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return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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ABufferSourceContext *abuffer = outlink->src->priv;
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outlink->sample_rate = abuffer->sample_rate;
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return 0;
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}
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static int request_frame(AVFilterLink *outlink)
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{
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ABufferSourceContext *abuffer = outlink->src->priv;
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AVFilterBufferRef *samplesref;
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if (!av_fifo_size(abuffer->fifo)) {
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av_log(outlink->src, AV_LOG_ERROR,
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"request_frame() called with no available frames!\n");
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return AVERROR(EINVAL);
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}
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av_fifo_generic_read(abuffer->fifo, &samplesref, sizeof(samplesref), NULL);
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avfilter_filter_samples(outlink, avfilter_ref_buffer(samplesref, ~0));
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avfilter_unref_buffer(samplesref);
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return 0;
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}
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static int poll_frame(AVFilterLink *outlink)
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{
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ABufferSourceContext *abuffer = outlink->src->priv;
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return av_fifo_size(abuffer->fifo)/sizeof(AVFilterBufferRef*);
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}
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AVFilter avfilter_asrc_abuffer = {
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.name = "abuffer",
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.description = NULL_IF_CONFIG_SMALL("Buffer audio frames, and make them accessible to the filterchain."),
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.priv_size = sizeof(ABufferSourceContext),
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.query_formats = query_formats,
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.init = init,
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.uninit = uninit,
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.inputs = (const AVFilterPad[]) {{ .name = NULL }},
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.outputs = (const AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.request_frame = request_frame,
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.poll_frame = poll_frame,
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.config_props = config_output, },
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{ .name = NULL}},
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};
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